[asterisk-dev] In call Codec parameterization in 1.6 ?

[Tim Panton]

On 1 Nov 2006, at 16:21, Paul Cadach wrote:

> Hello,
>
> Echo cancellation isn't job for codec, which should be ALWAYS  
> enabled at user's side, but softphones usually don't
> implements acoustic echo cancellers. To resolve echo problem you  
> should use some sort of headphones instead of
> "handsfree" operations (using microphone and speaker).

Oh, yes I know that's _best_ but unfortunately we have a picture of a  
girl talking to her laptop
on our website and it would be nice if that were practical :-)

Yes the canceler should run at the user side - but the user doesn't  
hear if it is needed or not,
the far-end does. There is _loads_ of cpu spare during the average  
voip call from a softphone,
why not use it for a good echo can - except that it will add to the  
latency - so we only want to
enable it if the user perceives the need for it.


>
> Choosing of right codec is main job for Asterisk and it is not  
> required any inband/outband interactions with client side
> (except for requesting to re-open media channel with right codec).  
> Currently codec selection isn't work very well and
> periodically causes some sort of problems, but it should be fixed  
> in near future (for example, check #4825 at
> bugs.digium.com).

I wasn't really thinking about negotiating the codec I was thinking  
of situations where the _parameters_
of the codec change during a call. AMR can come in multiple bitrates  
as can many video codecs, I'm
interested in supporting that sort of change during a call, not in  
choosing between Ulaw to GSM
during call set-up.

Tim Panton

www.mexuar.net